[Asterisk-Dev] How are SIP calls connected/bridged ?

[Arnaud]

Thanks Olle for the lecture. attempt_transfer() in chan_sip.c seems
like a good place to start digging.
- Arnaud

> So by reading chan_sip, you will not discover much, you need to dig
> deeper into the channel and pbx interface. In chan_sip, you can check
> the code for transfers that handle all these channels and redirecting
> them in various ways.